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deb1b1bc70
We should avoid creating codecs with invalid parameters, since this can expose security issues. For many codecs the IsOk() method to check the codec config is only called in DCHECKs. This CL ensures IsOk() is always called, also in non-debug builds. Bug: chromium:1265806 Change-Id: Ibd3c6c65d3bb547cd2603e11808ac40ac693a8b1 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238801 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35422}
87 lines
2.5 KiB
C++
87 lines
2.5 KiB
C++
/*
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* Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "api/audio_codecs/ilbc/audio_encoder_ilbc.h"
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#include <memory>
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#include <vector>
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#include "absl/strings/match.h"
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#include "modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h"
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#include "rtc_base/numerics/safe_conversions.h"
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#include "rtc_base/numerics/safe_minmax.h"
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#include "rtc_base/string_to_number.h"
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namespace webrtc {
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namespace {
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int GetIlbcBitrate(int ptime) {
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switch (ptime) {
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case 20:
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case 40:
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// 38 bytes per frame of 20 ms => 15200 bits/s.
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return 15200;
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case 30:
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case 60:
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// 50 bytes per frame of 30 ms => (approx) 13333 bits/s.
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return 13333;
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default:
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RTC_CHECK_NOTREACHED();
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}
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}
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} // namespace
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absl::optional<AudioEncoderIlbcConfig> AudioEncoderIlbc::SdpToConfig(
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const SdpAudioFormat& format) {
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if (!absl::EqualsIgnoreCase(format.name.c_str(), "ILBC") ||
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format.clockrate_hz != 8000 || format.num_channels != 1) {
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return absl::nullopt;
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}
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AudioEncoderIlbcConfig config;
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auto ptime_iter = format.parameters.find("ptime");
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if (ptime_iter != format.parameters.end()) {
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auto ptime = rtc::StringToNumber<int>(ptime_iter->second);
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if (ptime && *ptime > 0) {
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const int whole_packets = *ptime / 10;
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config.frame_size_ms = rtc::SafeClamp<int>(whole_packets * 10, 20, 60);
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}
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}
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if (!config.IsOk()) {
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RTC_DCHECK_NOTREACHED();
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return absl::nullopt;
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}
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return config;
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}
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void AudioEncoderIlbc::AppendSupportedEncoders(
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std::vector<AudioCodecSpec>* specs) {
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const SdpAudioFormat fmt = {"ILBC", 8000, 1};
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const AudioCodecInfo info = QueryAudioEncoder(*SdpToConfig(fmt));
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specs->push_back({fmt, info});
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}
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AudioCodecInfo AudioEncoderIlbc::QueryAudioEncoder(
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const AudioEncoderIlbcConfig& config) {
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RTC_DCHECK(config.IsOk());
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return {8000, 1, GetIlbcBitrate(config.frame_size_ms)};
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}
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std::unique_ptr<AudioEncoder> AudioEncoderIlbc::MakeAudioEncoder(
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const AudioEncoderIlbcConfig& config,
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int payload_type,
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absl::optional<AudioCodecPairId> /*codec_pair_id*/) {
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if (!config.IsOk()) {
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RTC_DCHECK_NOTREACHED();
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return nullptr;
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}
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return std::make_unique<AudioEncoderIlbcImpl>(config, payload_type);
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}
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} // namespace webrtc
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